Re: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP
Thanks Joshua. Submitted issue ASTERISK-23539 with the information. I verified my pjproject is up to date and included the latest git log commit I have just in case. Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, March 25, 2014 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP Dan Cropp wrote: I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I'm not seeing anything in the messages log. I'm sure I'm doing something wrong, just not sure where to look or how to track down the problem. It certainly shouldn't crash no matter what you do. Can you get a backtrace[1] and file an issue[2] so we can take care of this? The information you've provided in this email would also be useful. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP
I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I'm not seeing anything in the messages log. I'm sure I'm doing something wrong, just not sure where to look or how to track down the problem. Can anyone offer some hints? - pjsip.conf - [transport-udp] type=transport protocol=udp bind=0.0.0.0 [7001] type=endpoint transport=transport-udp context=IS disallow=all allow=ulaw auth=7001 aors=7001 [7001] type=aor max_contacts=1 contact=sip:7001@192.168.9.142:5063; Line 4 on my phone is setup for port 5063. ; I have also tried without this setting and am seeing the exact same scenario [7001] type=auth auth_type=userpass password=1234 username=7001 - extensions.conf - [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=DAHDI/G2 ; Trunk interface TRUNKMSD=1 [IS] exten = 1,1,Verbose(1,Unrouted call handler) exten = 1,n,Answer() exten = 1,n,Wait(1) exten = 1,n,Playback(tt-weasels) exten = 1,n,Hangup() Have a great day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP
=13d5988e59a920a6,qop=auth,nc=0001,cnonce=9adbf5ea Contact: 7001 sip:7001@192.168.9.142:5063 Expires: 240 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 8644 8644 IN IP4 192.168.9.142 s=- c=IN IP4 192.168.9.142 t=0 0 m=audio 16462 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv - Asterisk 12.1.1 Crashes at this point - From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Tuesday, March 25, 2014 4:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I'm not seeing anything in the messages log. I'm sure I'm doing something wrong, just not sure where to look or how to track down the problem. Can anyone offer some hints? - pjsip.conf - [transport-udp] type=transport protocol=udp bind=0.0.0.0 [7001] type=endpoint transport=transport-udp context=IS disallow=all allow=ulaw auth=7001 aors=7001 [7001] type=aor max_contacts=1 contact=sip:7001@192.168.9.142:5063; Line 4 on my phone is setup for port 5063. ; I have also tried without this setting and am seeing the exact same scenario [7001] type=auth auth_type=userpass password=1234 username=7001 - extensions.conf - [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=DAHDI/G2 ; Trunk interface TRUNKMSD=1 [IS] exten = 1,1,Verbose(1,Unrouted call handler) exten = 1,n,Answer() exten = 1,n,Wait(1) exten = 1,n,Playback(tt-weasels) exten = 1,n,Hangup() Have a great day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP
Dan Cropp wrote: I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I’m not seeing anything in the messages log. I’m sure I’m doing something wrong, just not sure where to look or how to track down the problem. It certainly shouldn't crash no matter what you do. Can you get a backtrace[1] and file an issue[2] so we can take care of this? The information you've provided in this email would also be useful. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
